FS: Asterisk Server Share

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Regs

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Snowflake

#199535 5-Mar-2009 22:59

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peewee101: SipX is a modular SIP proxy server that claims to be very stable, scalable, and reliable.
<snip snip snip>

Yeah, all the features are good, but how does it handle sip tunk connectivity? Thats my question. Does the RTP stream flow between your internal sip endpoints and the service provider's media server? In the case of a SIP trunk delivered over a private vlan or vpn its not usually something you would route on your LAN. In the Asterisk environment the RTP streams are between sip endpoints and the Asterisk server, then seperate RTP Streams from the Asterisk server to the Service providers SIP Media server. This eliminates the need to route between the LAN and private voice v-lan.

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Snowflake
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Regs

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Snowflake

#199537 5-Mar-2009 23:11

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Regs:
peewee101: SipX is a modular SIP proxy server that claims to be very stable, scalable, and reliable.

Yeah, all the features are good, but how does it handle sip tunk connectivity? Thats my question. Does the RTP stream flow between your internal sip endpoints and the service provider's media server? In the case of a SIP trunk delivered over a private vlan or vpn its not usually something you would route on your LAN. In the Asterisk environment the RTP streams are between sip endpoints and the Asterisk server, then seperate RTP Streams from the Asterisk server to the Service providers SIP Media server. This eliminates the need to route between the LAN and private voice v-lan.

http://www.sipfoundry.org/sip-trunking-interop-program.html

sipXecs is adding native SIP trunking support. Release 4.0 of sipXecs, due out early 2009, will include a complete and fully featured SIP trunking solution

Instead of using a traditional service provider to connect to the telephone network, you send all your calls over IP to an Internet Service Provider (ITSP). This saves you the cost of a PSTN gateway and instead you only need a data connection, which you most likely already have. Typically there is no difference in voice quality, however, you have to make sure that there is sufficient bandwidth available on the WAN side to accommodate all your voice calls and data traffic. It helps if your router or modem can assign quality of service priority to packets that carry voice traffic. sipXecs allows you to traverse local Network Address Translatoin (NAT) firewalls and makes sure all signalling and media is getting through. All you have to do is open certain ports on your firewall and direct traffic on these ports to sipXecs. sipXecs takes care of security and handles all that traffic approprietly

So it appears that the answer is "No", but capability is on the way.

Sales Engineer
Snowflake
www.snowflake.com

about.me/nzregs
Twitter: @nzregs

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